sr-users December 2018

sr-users@lists.kamailio.org

60 participants
73 discussions

Kamailio World 2019 - Call for Presentations

by Daniel-Constantin Mierla

Hello, I would like to announce that Call for Presentations at Kamailio World 2019 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k07/call-for-speakers/ The 7th edition of the event takes place again in Berlin, Germany, during May 6-8, 2019. Expect over 150 participants, developers and community members as well as representatives from other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training -- www.asipto.com

4 years, 6 months

SIP SIMPLE TO XMPP AND VICE VERSA

by sagar malam

Hello, We have a system which using FreeSWITCH for calls and Ejabberd( XMPP) for chat.Now we want to support SIP SIMPLE as well.But we are not going to move old users to SIP SIMPLE but only new users will use SIP SIMPLE.This leads to requirement of having SIP-XMPP gateway for which are are considering Kamailio server. I have gone through documentation about Kamailio XMPP module, it is clear that Kamailio can convert SIP MESSAGE to XMPP so SIP to XMPP will work but what about XMPP to SIP MESSAGE ? Does that work as well ? Thanks in advance. -- Thanks, Sagar

4 years, 6 months

Dialplan Module limits

by Enrico Bandiera

Hi guys, does anyone know which are the limits of the diaplan module? After how many entries I will encounter serious performance issues? Thanks, Enrico.

4 years, 6 months

question about replace_body_all

by Wilkins, Steve

Hello, I am attempting to change the Message Body of an SIP message using replace_body_all; I want to replace all attributes that start with rtcp. Here is my code => replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call msg_apply_changes() after this call. Note that I verified that the regex is working using search. The problem is that nothing gets changed. However, what I do get is a new string at the bottom of the Message Body with my newstring concatenated three times, which happens to be the number of times that rtcp occurs in the Message Body. Note, I am doing this so that I can add the IP address to the rtcp attribute because this is required by a Provider. Any ideas? Thank you, -Steve

4 years, 6 months

Dialog getting disconnected as soon as call is ended

by Subham Singhal

Hello, I am new to kamailio. I have been trying to integrate kamailio with asterisk. I am able to send INVITE and get RTP packets as well, but I am unable to receive BYE from asterisk. Call is getting terminated through following websocket error. "WebSocket connection to 'wss://172.16.16.169:5061/' failed: One or more reserved bits are on: reserved1 = 1, reserved2 = 0, reserved3 = 0." -- Thanks and Regards Subham Singhal

4 years, 6 months

Auth_xkeys module - freeing already freed pointer

by José Seabra

Hello there, I'm constantly getting the following message when i'm using module auth_xkeys. CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7f6310987400), called from core: core/data_lump.c: free_lump(466), first free core: core/data_lump.c: free_lump(466) - ignoring Apparently kamailio only throw this message but i would like to know if its a normal protection from kamailio or it should be fixed? Thank you for the support Regards -- José Seabra

4 years, 6 months

Assistance with re.subst & newline

by Mark Hall

Hi, I am using rtpengine to proxy audio to/from media servers. The outbound calls are being rejected by a couple of my carriers (although accepted by others), and the closest that I have been able to find for the reason is the inclusion of "a=rtcp:xxxxx" in the SDP generated by rtpengine. To test further, I am trying to remove this line. Rtpengine itself doesn't have any options to remove it, so I am left with manipulating the generated SDP in Kamailio. A sample of generated SDP is: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:30939 s=My Media Server I am using rtpengine_manage("RTP/AVP ICE=remove"); to generate the SDP to a variable ($avp(sdp)) Now, I am trying to just remove the whole a=rtcp line with re.subst $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*\R||s}); *** Tried / as delimiter also, but | looks cleaner *** This should match the line and any CR/LF characters - BUT in results, nothing is changed (various online Regex testers show this match to be correct - I tried \r\n also). If I do the same but without the \R, then the line is removed but not the CR/LF, leaving a gap in the SDP - which is invalid. $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*||s}); Gives: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv s=My Media Server Can anyone see where I am going wrong, or suggest an alternative? Kind Regards, Mark Hall

4 years, 6 months

New features: dialog - list dialogs via rpc using matching expressions

by Daniel-Constantin Mierla

Hello, being discussed and requested couple of times via lists, irc channel and discussions at events world wide, I just want to send a short note to announce that yesterday I pushed a new rpc command to dialog module that returns dialog records that match on conditions applied to request uri, from uri, to uri or callid fields. The filtering can be done with string comparison, regular expression or starts-with (prefix) matching. See more details at: - https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.list_… It should enable listing dialogs only for specific users, caller or callee, without dumping all dialogs. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

4 years, 6 months

set_advertised_address

by Kjeld Flarup

Hello I have a PBX behind NAT. Thus I advertise the public IP, and forwards the port to my PBX. listen=LOCALIP:5070 advertise EXTERNALIP:5070 Now clients can connect to the PBX from the Internet. And also inside the LAN, because I have enabled NAT loopback. However some customers sysadmins complains that NAT loopback is a security risk. I have not been able to find any exploits of this, but the sales and support people asks if it is possible to remove this NAT loopback requirement. I could look at $rd and if it is local, then I could advertise LOCALIP. I found set_advertised_address("LOCALIP"); set_advertised_address however only seems to modify the latest Via header, not the Record-route, and audio neither works. Could I do something to make this work, or is it a dead end? -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 years, 6 months

Dispatcher clear dst_avp

by Grant Bagdasarian

Hello, I'm trying to use multiple destination sets which may include the same destination. For instance: SET ID: 1 sip:10.0.0.1:5060 sip:10.0.0.2:5060 SET ID: 2 sip:10.0.0.1:5060 Whenever a new SIP INVITE is sent to the Kamailio instance it will use SET ID 1. If then for some reason this call attempt fails, failure_route will trigger and SET ID 2 will be used. At first glance this might not make any sense, but there is some business logic behind this, but it's not relevant for this case. For some reason, the dst_avp variable still contains the destinations of SET ID 1 in failure_route. Is there a built-in way to reset this variable for every call to ds_select_domain(), such that only the destinations of passed SET ID will be loaded into the avp? I tried setting the avp to $null before calling ds_select_domain, but that had some undesirable effects. The reason I'm asking this is the destinations the call is being sent to are Freeswitch instances, which will return a 482 Request Merged when the same INVITE is sent to the same Freeswitch within 4 seconds (default T4 timer). I'm trying to match the previously selected destination in failure_route with the next destination and skip if they're the same, or stop processing when they are the same and no more destinations are available. I know I can mess with the T4 timer in Freeswitch and perhaps even sleep in failure route to deal with the 482 failure, but I rather fix this in a normal way. Regards, Grant Bagdasarian Senior Developer

4 years, 6 months

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sr-users December 2018